Displaying 20 results from an estimated 3000 matches similar to: "Call go on hold for no reason"
2006 May 09
0
Using ChanIsAvail and SIP
I am trouble finding a configuration that works for ChanIsAvail and SIP.
My two providers are Voxee and Teliax.
I have these lines in a macro
exten => s,n,ChanIsAvail(SIP/teliax&SIP/voxee)
exten => s,n,Cut(CH=AVAILCHAN,-,1)
exten => s,n,NoOp(AVAILCHAN= ${CH})
; Dial the available Channel
exten => s,n,Dial(${CH}/${ARG1},60,t)
Looking at the execution, I can see what the AVAILCHAN
2006 Jan 22
1
Fail over using CHANAVAIL
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but this returns the provider but not the username, so I
don't understand how to use this for real
2006 Jan 28
0
AutoDialing with VOP USING SIPURA 2100'S
Hello all,
I am trying to find out if anyone has a provider that is good with dtmf
playback using a Sipura 2100? I've just dialed with voxee and the call goes
through but when I press 1 my dialer does not " hear" it.
My dialer is making the call using a Dialogic d/4PCI connected to the
Sipura 2100 through voxee and I am calling my landline. When I pick up the
landline
2003 Oct 05
1
ChanIsAvail app setting ${AVAILCHAN} to an unusable value.
I sent this earlier under "Editting variable contents" but no-one
has responded. So, the subject is now more to the problem, instead of
the solution I was trying to implement.
ChanIsAvail returns the channel ID plus "-<session>".
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a
2007 Jan 24
1
ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro.
The logic is simple; try Zap/g1 (a group of two E1s), and if that
fails, try locating a channel via DUNDi. Here's a massively cut down version
to illustrate the problem I'm having.
macro dialout ( dest ) {
ChanIsAvail(Zap/g1);
noop(Value of AVAILCHAN is ${AVAILCHAN});
2004 Jul 18
0
ChanIsAvail issue
Hello
I am trying to setup ChanIsAvail function in the extensions.conf file so that user should use the available channel to call out, but immediately after the function like, zap channel hangup.
Here is the copy of my extensions.conf file and messages display on consol while making the call.
Please help me to fingure out this issue.
Thanks
Deepak
Extension.conf :
exten =>
2006 Oct 13
2
AEL Question
Hi, all. I'm making my first foray into AEL. I'm starting with a
simple macro, but I've already encountered an odd behaviour. I'm
wondering if someone can shed some insight:
Asterisk 1.2.9.1
macro newPlaceCallPSTN {
s => {
TIMEOUT(absolute)=7200;
NoOp(Analog Out List: ${ANALOGOUT});
ChanIsAvail(${ANALOGOUT});
NoOp(Available Out List:
2005 Jun 27
1
Re: teliax [Was: LiveVoip is Bankrupt]
For outbound only, I have traditionally recommended VoipJet. They just
recently has a spat of issues that seem to have resolved though. I am
still using them via their east coast server and it seems to work quite
well again. Cost is around 1.3 cents minute I believe. Use IAX and
g711 for best quality to VoipJet.
Thanks,
Wiley
-----Original Message-----
From:
2005 Jan 27
1
ChanIsAvail not working
I'm testing ChanIsAvail context and it is not working for me.
exten => 55,1,ChanIsAvail(SIP/11&SIP/21)
exten => 55,2,Cut(theChannel=AVAILCHAN,,1)
exten => 55,3,Dial(${theChannel},r)
exten => 55,4,Hangup
exten => 55,102,Goto(s,4)
It is not dialing SIP/21 when I'm talking on SIP/11, it execute
Hangup instruction instruction.
According to notes:
The channels are checked
2004 Apr 08
1
Two operators, 10 rollover lines, Cisco 7960G chanisavail problem
Here's my situation.
I have two receptionists that answer incoming lines. Each has a 7960G with
5 incoming lines each. I'm trying to set this up so each line on each phone
doesn't utilize call waiting. My problem seems to be that
ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always
returns cisco1.
Here are the sip.conf entries: (mind you,
2003 Oct 03
1
Editting variable contents
ChanIsAvail returns the channel ID plus "-<session>".
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a warning, but dialing a SIP channel
completely errors out.
------ extensions.conf snippet-------------
;
; Main Home number (8901)
;
; Bedroom1
exten =>
2006 Feb 14
4
ChanIsAvail
Hi,
So I've done my research on Chanisavail, read the wiki, checked the
archive but can't seem to find anything to suit my scenario. I've
played around with it a lot, but I'm still scratching my head on what
I need to do.
What I need is to be able to accept a call by SIP and ring all
telephones that are not in use (which just so happen to be on Zap
interfaces, but might be SIP
2005 Jun 30
0
Re: Asterisk-Users Digest, Vol 11, Issue 181
Hi,
I am new to asterisk , i am getting the following
error,& the /etc/zaptel.conf file entry is as follows
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs,yellow
bchan=1-23
dchan=24
Parsing '/etc/asterisk/zapata.conf': Found
Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664
zt_open: Unable to specify channel 1: No such device
or address
Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296
2007 Feb 09
0
Conference & Page question
Hi. I'm currently setting up a particular conference: 3 members (a,b,c),
a can speak with b and c, b and c can speak only with a and not between
them.
I found my possible solution with paging/intercom using option "d"
(full-duplex), but I need to make ringing the phone in intercom.
Now I set auto-answer=6 but after first ring the phone hangup the call.
There is a way to using
2023 Sep 08
1
Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Some progress to report:
I had to run asterisk as the user logged in - actually not even that. I
could not "su user -c " to that user - I had to run it as that user.
Then I did a test and got audio! Great...
But when I do a second test. Asterisk HANGS on ChanIsAvail()
Then I thought lets SKIP that - and just let it do the Dial() - I stopped
everything - got it running again. - and
2008 Feb 20
3
Dial+Macro and Queue
A call comes in and goes into the queue, the queue dials a sip channel using
a macro. The macro plays a set of options to the callee and if the callee
presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason
the caller goes back into the queue rather than continueing on in the dial
plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE
that the
2005 Jun 20
1
sipredirect question
Hi all,
I want to build a central call diverter via asterisk (http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SIPredirect). Calls would come in via SIP from Cisco Callmanager Asterisk would do some searching an diverts the call to an extension, which is also located at Callmanager.
When like to use >>sipredirect<< Asterisk complains "No application
2014 Jan 13
0
How to get ringing sound in outbound call in asterisk
I have two server
Server_A(outbound call) for agent login and agent make a outbound call from
here and pass into server Server_B call
extension.conf
exten => _91XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR)
exten => _91XX.,n,hangup()
Server_B[192.168.53.197] for call forwarding
extension.conf
exten =>
2004 Aug 20
3
Strange problem with Dial
I'm trying to add an emergency dial to my context. However, when I try to
dial it, I get caught in an endless loop.
For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called. Using two different extensions to call teh
same number, I get two different actions by *.
Here is the vvverbose output:
-- Starting simple switch on
2007 Jun 25
1
Problems with ChanIsAvail always return status 0
Hi list:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
THanks.
In the example i'm dialing from extension SIP/112
My DialPlan Secction:
[macro-callonlyiffree]
exten => s,1,ChanIsAvail(${ARG1}|s)
exten => s,n,NoOp(${AVAILCHAN})
exten